ty.roach@acecomm.com
2004-Nov-05 11:45 UTC
[Asterisk-Users] Questions from an Asterisk newbie
I have just installed asterisk in the hopes of operating a very simple VoIP demo. The demo environment is as follows: Asterisk 1.0.2 installed on a Fedora 2 Linux laptop. The laptop is connected to a hub along with two Cisco 7960 IP phones (SIP enabled). I've manually configured the phones setting the IP address of the phones, phone names (extensions), the IP address of the SIP proxy (Asterisk server?). I have not made any modifications to any of the asterisk configuration files. I run asterisk ('asterisk -cv') from the command line just to see what happens. Essentially, I get messages from both SIP phones indicating that registration is failing (I guess not such as surprise since I haven't configured anything). For starters, I was hoping that some of the experts on this board could give me some tips on what I need to do to allow one phone to successfully call the other phone. I did a similar thing several years ago using a SIP proxy server (from Dynamicsoft, albeit, with help from their support group). Any advise would be greatly appreciated. Thanks and advance. Ty P.S. I've included command line output from my asterisk console below... *CLI> sip debug SIP Debugging Enabled *CLI> *CLI> *CLI> *CLI> Sip read: REGISTER sip:172.20.23.201 SIP/2.0 Via: SIP/2.0/UDP 172.20.23.211:5060 From: sip:4444@172.20.23.201 To: sip:4444@172.20.23.201 Call-ID: ce30300-411dcd5-8f0953-2e323731@172.20.23.211 CSeq: 101 REGISTER Contact: <sip:4444@172.20.23.211:5060> Expires: 3600 Content-Length: 0 9 headers, 0 lines Using latest request as basis request Sending to 172.20.23.211 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 172.20.23.211:5060 From: sip:4444@172.20.23.201 To: sip:4444@172.20.23.201;tag=as106566ef Call-ID: ce30300-411dcd5-8f0953-2e323731@172.20.23.211 CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:4444@172.20.23.201> Content-Length: 0 to 172.20.23.211:5060 Nov 5 13:37:01 NOTICE[-159417424]: chan_sip.c:7571 handle_request: Registration from 'sip:4444@172.20.23.201' failed for '172.20.23.211' Scheduling destruction of call 'ce30300-411dcd5-8f0953-2e323731@172.20.23.211' in 15000 ms Destroying call 'ce30300-411dcd5-8f0953-2e323731@172.20.23.211' Sip read: REGISTER sip:172.20.23.201 SIP/2.0 Via: SIP/2.0/UDP 172.20.23.212:5060 From: sip:3005@172.20.23.201 To: sip:3005@172.20.23.201 Call-ID: 2ae30300-4302418-8f1c2b-2e323731@172.20.23.212 Date: Fri, 05 Nov 2004 18:38:54 GMT CSeq: 101 REGISTER Contact: <sip:3005@172.20.23.212:5060> Expires: 3600 Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 172.20.23.212 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 172.20.23.212:5060 From: sip:3005@172.20.23.201 To: sip:3005@172.20.23.201;tag=as66d562fd Call-ID: 2ae30300-4302418-8f1c2b-2e323731@172.20.23.212 CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3005@172.20.23.201> Content-Length: 0 to 172.20.23.212:5060 Nov 5 13:37:36 NOTICE[-159417424]: chan_sip.c:7571 handle_request: Registration from 'sip:3005@172.20.23.201' failed for '172.20.23.212' Scheduling destruction of call '2ae30300-4302418-8f1c2b-2e323731@172.20.23.212' in 15000 ms
Jean-Michel Hiver
2004-Nov-05 12:22 UTC
[Asterisk-Users] Questions from an Asterisk newbie
Hi,>For starters, I was hoping that some of the experts on this board could >give me some tips on what I need to do to allow one phone to successfully >call the other phone. I did a similar thing several years ago using a SIP >proxy server (from Dynamicsoft, albeit, with help from their support >group). >I am a n00b too - Yet I had some limited success using this introduction: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html Cheers, Jean-Michel.
HI, First you should start by editing two different .conf files: sip.conf and extensions.conf in /etc/asterisk The best way is to let asterisk create the samples command make samples in /sr/src/asterisk Sample .conf files will be created in /etc/asterisk A step by step info can be found in http://www.voip-info.org/wiki-Asterisk+installation+tips go to the part of SIP configuration. There is also some info about your Cisco phones. Good Luck Paulo Francisco Paulo Adriano WaveLIS LDA Mobile +351 91 870 87 98 Office + 351 21 989 83 34 Fax +351 21 989 83 35 E-mail : pauloadriano@wavelis.pt>>> ty.roach@acecomm.com 05-11-2004 18:45:46 >>>I have just installed asterisk in the hopes of operating a very simple VoIP demo. The demo environment is as follows: Asterisk 1.0.2 installed on a Fedora 2 Linux laptop. The laptop is connected to a hub along wittwo Cisco 7960 IP phones (SIP enabled). I've manually configured the phones setting the IP address of the phones, phone names (extensions), the IP address of the SIP proxy (Asterisk server?). I have not made any modifications to any of the asterisk configuration files. I run asterisk ('asterisk -cv') from the command line just to see what happens. Essentially, I get messages from both SIP phones indicating that registration is failing (I guess not such as surprise since I haven't configured anything). For starters, I was hoping that some of the experts on this board could give me some tips on what I need to do to allow one phone to successfully call the other phone. I did a similar thing several years ago using a SIP proxy server (from Dynamicsoft, albeit, with help from their support group). Any advise would be greatly appreciated. Thanks and advance. Ty P.S. I've included command line output from my asterisk console below... *CLI> sip debug SIP Debugging Enabled *CLI> *CLI> *CLI> *CLI> Sip read: REGISTER sip:172.20.23.201 SIP/2.0 Via: SIP/2.0/UDP 172.20.23.211:5060 From: sip:4444@172.20.23.201 To: sip:4444@172.20.23.201 Call-ID: ce30300-411dcd5-8f0953-2e323731@172.20.23.211 CSeq: 101 REGISTER Contact: <sip:4444@172.20.23.211:5060> Expires: 3600 Content-Length: 0 9 headers, 0 lines Using latest request as basis request Sending to 172.20.23.211 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 172.20.23.211:5060 From: sip:4444@172.20.23.201 To: sip:4444@172.20.23.201;tag=as106566ef Call-ID: ce30300-411dcd5-8f0953-2e323731@172.20.23.211 CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:4444@172.20.23.201> Content-Length: 0 to 172.20.23.211:5060 Nov 5 13:37:01 NOTICE[-159417424]: chan_sip.c:7571 handle_request: Registration from 'sip:4444@172.20.23.201' failed for '172.20.23.211' Scheduling destruction of call 'ce30300-411dcd5-8f0953-2e323731@172.20.23.211' in 15000 ms Destroying call 'ce30300-411dcd5-8f0953-2e323731@172.20.23.211' Sip read: REGISTER sip:172.20.23.201 SIP/2.0 Via: SIP/2.0/UDP 172.20.23.212:5060 From: sip:3005@172.20.23.201 To: sip:3005@172.20.23.201 Call-ID: 2ae30300-4302418-8f1c2b-2e323731@172.20.23.212 Date: Fri, 05 Nov 2004 18:38:54 GMT CSeq: 101 REGISTER Contact: <sip:3005@172.20.23.212:5060> Expires: 3600 Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 172.20.23.212 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 172.20.23.212:5060 From: sip:3005@172.20.23.201 To: sip:3005@172.20.23.201;tag=as66d562fd Call-ID: 2ae30300-4302418-8f1c2b-2e323731@172.20.23.212 CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3005@172.20.23.201> Content-Length: 0 to 172.20.23.212:5060 Nov 5 13:37:36 NOTICE[-159417424]: chan_sip.c:7571 handle_request: Registration from 'sip:3005@172.20.23.201' failed for '172.20.23.212' Scheduling destruction of call '2ae30300-4302418-8f1c2b-2e323731@172.20.23.212' in 15000 ms _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041105/43decd57/attachment.htm
ty.roach@acecomm.com wrote:> I have just installed asterisk in the hopes of operating a very simple VoIP > demo. The demo environment is as follows: > > Asterisk 1.0.2 installed on a Fedora 2 Linux laptop. The laptop is > connected to a hub along with two Cisco 7960 IP phones (SIP enabled). I've > manually configured the phones setting the IP address of the phones, phone > names (extensions), the IP address of the SIP proxy (Asterisk server?). > > I have not made any modifications to any of the asterisk configuration > files.I just did something similar. I added these lines to /etc/asterisk/sip.conf: ; Grandstream [1001] type=friend host=dynamic ; cisco phone [1002] type=friend host=dynamic Then I added these lines to /etc/asterisk/extensions.conf exten => 1001,1,Dial(SIP/1001,200,tr) exten => 1002,1,Dial(SIP/1002,200,tr) My phones register as phone numbers 1001 and 1002. There may be a better way to do it, but with this config I was able to make calls... Ben -- Ben Greear <greearb@candelatech.com> Candela Technologies Inc http://www.candelatech.com